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Summary:ASTERISK-27844: Any changes in sip.conf and then sip reload causes all call drops in case of registering sip as a client to tollfree providers
Reporter:Abhinav Nimesh (Abhinav)Labels:
Date Opened:2018-05-08 14:26:05Date Closed:2020-01-14 11:14:04.000-0600
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General Channels/chan_sip/Registration
Versions:13.16.0 Frequency of
Occurrence
Frequent
Related
Issues:
Environment:Cent os 7, 16 GB RAM, 1tb hardiskAttachments:
Description:When do sip reload from asterisk cli Everything is fine but when I make some change in sip.conf (even a small change just adding one comment statement) and then reloading causes all call drops. Even if I just do touch sip.conf ( anything that changes it's time stamp) and than reloading causes all call drops.
Note: This is happening only in case of registering asterisk to tollfree providers ( using register line in general section ).
I am using asterisk13.
Comments:By: Asterisk Team (asteriskteam) 2018-05-08 14:26:06.134-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

By: Joshua C. Colp (jcolp) 2018-05-08 14:29:07.331-0500

It appears the bug you have submitted is against a rather old version of a supported branch of Asterisk. There have been many issues fixed between the version you are using and the current version of your branch. Please test with the latest version in your Asterisk branch and report whether the issue persists.

Please see the Asterisk Versions [1] wiki page for info on which versions of Asterisk are supported.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions



By: Abhinav Nimesh (Abhinav) 2018-05-10 06:51:18.619-0500

the issue still persist with asterisk 13.21, Using wireshark i found that the BYE message is initiated from the provider side (does i have to ask my provider why they doing this?)

By: Joshua C. Colp (jcolp) 2018-05-15 02:46:59.564-0500

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. The specific steps or actions you took that caused you to encounter the problem.
2. The behavior you expected and the location of documentation that led you to that expectation.
3. The behavior you actually encountered.

To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

[2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



By: Abhinav Nimesh (Abhinav) 2018-05-25 01:27:00.192-0500

; Date: Thu May 03 20:05:54 IST 2018
; Voice Resource Id: 1
[general]
bindport=5060
bindaddr=0.0.0.0
sendrecvHeaderDisabled=no
silenceSuppHeaderDisabled=no

context=from-manager-core-incoming
allowguest=no

srvlookup=yes


rtptimeout=300
rtpholdtimeout=600

disallow=all
allow=g729
allow=alaw
allow=ulaw

register =>xxx:xxx@xx.x.xx.xxx/xxx
defaultexpiry=300
qualify=no


[tatasip]
type=friend
username=xxx
secret=
host=xx.x.xx.xxx
port=5060
dtmfmode=rfc2833
fromdomain=xx.x.xx.xxx
nat=force_rport,comedia
canreinvite=no
context=extension-context-ASTERISK_SIP-8
;Custom Configuration
;context=from-tatasip
insecure=invite,port
fromuser=xxx
fromname=xxx
setvar=SIPADDHEADER01=P-Preferred-Identity: <sip:xxx@xx.x.xx.xxx>

I am using asterisk as a sip client.
open sip.conf
make any changes in it, just add a semicolon and save it.
Now from AsteriskCli do sip reload -> this lead to all call drops...


By: Abhinav Nimesh (Abhinav) 2018-05-25 02:37:07.452-0500

After analyzing TCP dump on Wireshark, I found that Bye is sent from Provider's IP.

By: Joshua C. Colp (jcolp) 2018-05-29 04:34:56.123-0500

If the upstream provider is the one sending the BYE then I'm not really sure what could be done Asterisk side to prevent this. It seems to be some sort of policy or decision on their side.

By: Asterisk Team (asteriskteam) 2018-06-12 12:00:01.593-0500

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines